similar to: Friday July 31st at 12 Noon EDT: Dave Nelsen, Skype for Asterisk beta opens, Gizmo Voice + Google Voice = free SIP calls

Hi, Plenty to talk about today. Dave Nelsen, a founder of Talkshoe, has a lot of experience in the telecommunications space and he joins us today to chat about its current state, conferencing and whatever else comes to mind. So we have a meta conference aout conferencing, it won't be the first time :) You probably saw John Todd's message on one of the lists: Skype for Asterisk is in open

Hi All, At the usual time, 12 Noon ET on Friday April 3rd, we expect Michael Robertson to join the discussion to filed questions about OpenSky and Gizmo5. I have been testing all of these Skype to X methods except SIP for Skype since I have no word from them. I can tell you that we've had good results with bith Skype for Asterisk and OpenSky. In fact, I am currently accepting calls to my

Anyone got any thoughts on this and how it compares to the chan_skype that's due soon ? "OpenSky is a free service provided by Gizmo5 which allows *any* mobile phone, web browser or IP aware phone network (SIP, asterisk, etc) to communicate with Skype users. OpenSky supports sending text messages and voice calls." http://www.gizmo5.com/pc/opensky/ Julian

>> On Feb 13, 2009, at 11:19 AM, Philipp von Klitzing wrote: >> >> > Hi there, >> > >> > is gizmo the first user of the Digium Skype solution, or do they use a >> > different approach/product - any clue? >> > >> > http://www.gizmo5.com/pc/opensky/ >> > >> > Philipp OpenSky is no related to any product from Digium.

Hi all, In about 4 hours from this writing, the G.722 conference bridge will be brought up, the Talkshoe G.711 also, so you can call in via SIP, PSTN or Skype (experimental) http://vuc.me for all the gritty details IRC #voip-users-conference anytime today We'll also be talking about hosted PBX systems and what they might need to bring telephone into not the 20th, but the 21st century. It

Daniel wrote: For us, opensky can be OK for individual users, not for allowing Asterisk users to call Skype users. Why? Simply that when you buy the 20 USD connection to Skype and don't want your calls to be cutted after 5 mn, you have to use the Gizmo Skype aliases system which is in your account. Not really helpful if you want to connect transparently your users to Skype! They better had to

Hi there, is gizmo the first user of the Digium Skype solution, or do they use a different approach/product - any clue? http://www.gizmo5.com/pc/opensky/ Philipp

> Anyone connected up to it yet? > > http://www.skypeforsip.com/ This service is vaporware. It's just surveyware at this point with no actual service. An alternative is OpenSky which is a launched service which does SIP to Skype and Skype to SIP so you can answer and make all your Skype calls from any SIP aware device. There's a comparison chart at: http://sipforskype.com and

Hi, This week's VoIP Users Conference will be tie the Talkshoe PSTN/SIP ULAW conference bridge to the ZipDX.com G.722-capable bridge. It may be a little crazy, but I'm looking forward to getting some explanations from David Frankel about the effects of wide band (or as Polycom calls it, "HD Voice") on conferencing. David can probably answer any technical questions you may have

The subject of tomorrow's VoIP Users Conference will be mobile VoIP. If you have any interest, please join us. I myself am tesing a bunch of iPod applications to use with all the usual suspects: OnSIP, Sipgate, Gizmo, Skype, your asterisk box, etc. Details for joining the call are are at http://VUC.me or http://VoipUsersConference.org IRC: #voip-users-conference on Freenode.net See you

I have written this configuration script which uses OpenSky to make Skype calls directly from Asterisk devices using my companies SIP to Skype gateway. Users can dial skype_anyskypeusername or manually add names or extensions which can get mapped to the correct dialing sequence. The right sequence is username at opensky.gizmo5.com but that gets mapped to sipphone address so I set that up to map

Tomorrow, Friday August 21st at noon EDT, we'll be on the rampage again with the VoIP Users Conference, much of which concerns Asterisk and the Asterisk community. We've covered a lot of ground in the past 2 1/2 years, and had a lot of great guest presenters. In the meantime, there's a new gadget to test at PhoneFromHere.com, you can hear the wideband demo on the ZipDX.com bridge:

Today, Chris Matthieu, Founder & CEO of GetVocal, entered the cloud-based communications market in February, with its launch of Teleku. Teleku is a new cloud-based telecom service that allows Web developers to build and host phone applications that answer inbound calls and initiate outbound calls, interact with Web applications, and send/receive SMS text messages! Chris, who you may have

Hi, Cory Andrews had a last minute travel plan change and even the Snom M3 won't help on a plane. We'll reschedule Cory. We're open to any and all asterisk-related ("asterisk" is a trademark of Digium) discussion or VoIP related discussion this week. If you're into VoIP, and you must be if you're on this list, please join us, we solve the world's VoIP problems

Friday March 28th;: Astlinux How to be a part of the conference via PSTN or SIP : http://www.VoipUsersConference.org Astlinux is a custom Linux distribution that is centered around the goal of providing Asterisk. Astlinux is highly optimized for Asterisk both in commercial and embedded systems. They will be releasing a new version shortly and Darrick Hartman has graciously offered to be a

Yes, we mostly talk about asterisk, hardware, software, phones, people, events and things asterisk-related. Asterisk is a registered trademark of Digium. Asterix is a registered trademark of Ren? Goscinny. We never discuss that, though. Here's the short URL for sending out to others that might be interested: http://x2z.eu It lands here: http://www.VoipUsersConference.org Once again, I

Morning! This may be the "day of the dead" in some regions, but we expect the usual lively discussion today at 9AM PDT, 11 Central, 12 Noon EDT, 4PM UK and Portugal, 5PM Paris, $deity-forsaken hour down under. This Sunday, I believe the USA falls back to Standard time. Future VUC are still at 12 Noon EST. Info site: http://voipusersconference.org PSTN (724) 444-7444 enter 22622# 1#

Hi, I met Michael Iedema of the Askozia pbx project at ANOOMA (ANOOMA is the Cannes Festival of the Asterisk world) and begged him to take time off from porting Askozia from FreeBSD to linux to tell us about his project and the problems of porting as well. Hope you'll all be able to stop by and lmisten or download or stream the recordings. We are having a lot of people call in on widenabd

Hi, Like me, some of you probably remember Jim as one of the pioneers along with Leif and Jarod. These guys "wrote the book", literally. Jim is our guest tomorrow and he'll be talking about system building, among other things. We always have a good time AND get stuff done on the Conference so come by and join us: Silently, on IRC: #voip-users-conference (freenode.net) - stop by

Already Friday, the week went by in a Flash. If you haven't yet registered for a free Sipgate DID, I suggest you go do so. If you are in, or interested in the business, Sipgate has a few tricks up their sleeves and you should be aware of them. Someone from Sipgate will be joining the conference, which begins at 12 Noon EDT (9AM PDT, 10 Mountain, 11 Central, 5PM UK, 6 Central Europe) ZipDX